I would like to send G729 codec to my softswitch.
I have pbxes pro account. When I making call from my pbxes to my softswitch it's sending GSM codec but If I make call from outside pbxes using another DID provider then I am getting correct codec G729 codec on my swtich.

Is there any way to pass G729 codec to my switch from pbxes instead of GSM.

• What kind of soft-switch are we talking about?
• Will the soft-switch terminate the inbound PBXes call, to the PSTN via one of its' own trunks?
• Can I assume that trunk supports the G.729 vocoder?
• Does the SIP UA initiating the call have the G.729 in its' SDP?

I ticked on G729 Passthrou under General settings but still It doesn't work.

Please check below answer.

• What kind of soft-switch are we talking about?
Ans : I have voipswitch from .voipswitch (dot) com
• Will the soft-switch terminate the inbound PBXes call, to the PSTN via one of its' own trunks?
Ans :I making inbound call from pbxes using DID which is registered with pbxes and it's sending call to my voipswitch using IVR but it's sending GSM codec instead of G729 - outbound call going from my voipswitch.

• Can I assume that trunk supports the G.729 vocoder?
Ans : I used one DID directly point to my IP address then it's sending G729 codec to my voipswtich. Same DID I registered and sending call from pbxes then it's sending Codec GSM to my softswitch. So GSM codec definaly coming from pbxes becasue I can call fine using DID if I directly point to my IP address.

• Does the SIP UA initiating the call have the G.729 in its' SDP?
Ans : Yes (I can connect fine if I directly point DID to IP then it's sending correct codec and I can make call successfully but I routing from pbxes because I need IVR)

Can you please advise me It will be good because I really like Pbxes services and I want to use it.

OK, let's run through this call scenario step by step. Please correct me if I am wrong, or if I omit a step.

An inbound call from the PSTN has arrived to your DID. The DID provider's SIP Proxy sends the call to the SIP URI voipswitchuser@PublicStaticIPofVoiPSwitch which in turn sends the call to a SIP UA which supports the G.729 vocoder. Apparently, this call scenario works as intended, with the G.729 vocoder being the common codec across all the call legs.

An inbound call from the PSTN has arrived to your DID. The DID provider's SIP Proxy sends the call to the SIP URI softphone-xxx@pbxes.org which in turn sends the call to the SIP URI voipswitchuser@PublicStaticIPofVoiPSwitch that in turn sends the call to a SIP UA which supports the G.729 vocoder. Apparently this call scenario doesn't work as intended, since the G.729 vocoder is not the common codec across all the call legs. The call leg from PBXes to the VoiPSwitch is set to the GSM codec for some reason.

Here is the test call setup you should use to troubleshoot this: The DID provider's SIP Proxy sends the call to the SIP URI softphone-xxx@pbxes.org which in turn sends the call to a PBXes extension that allows you to look at which vocoder has been chosen during the call. It is quite easy to see the vocoder on the Linksys SPAxxx and on the Snom phones, or you can setup a packet trace with any other SIP UA that doesn't indicate the vocoder during the call.

If after a few test calls, the vocoder happens to be the G.729 all the way through to the extension, then you will have verified the PBXes' G.729 Passthru feature works properly with this particular DID. Then set the Inbound Route to send the call through to an IVR before it sends it to the same extension. If the vocoder is GSM instead of G.729 then the IVR creates the transcoding issue.

Note: Snom phones are best for these kind of tests, since they support both the GSM as well as the G.729 vocoders, and their built-in SIP trace can be used as a reference.

I tested three different providers DID with pbxes but it's only sending GSM codec.

If I will pay for personal support then your support team will help me regarding this issue.? Is it possible all calls from my account it will use G729 codec instead of GSM? If you have any alternative please let me know. Thanks.

If you tested this with three different SIP Proxies, then chances are the PBXes SIP Proxy is creating the issue. Please open a support case, where we will be able to exchange some information, and look more closely at this issue.

One of our voip provider (NEAGEN) requires G729 only. We are considering to alingning all our clients (phones and softphones) to G729 only. We connect some of our clients (usually voipphones Diemens Gigaset plus some Linksys or rarely softphones) to Pbxes and, via Pbxes, we make calls using Cheapnet or Neagen (Voip providers, calls to Italian DIDs, or we receive call on DIDs associated to their voip accounts, calls mad by regular landlines or mobiles.

We read this on your shop section: "G.729 Compression - An option for low bandwidth requirements. Premium Accounts already support G.729 in "Passthru" mode which does not allow putting calls on hold (including most transfers) or recording. This option provides full support with transcoding"

Should we buy the license for G729 in you shop at 2.95 Euro per month? When we try to buy this option we are requested to fill a box with the number of channels ("massimo canali:......") Please let us know this matter.

Should we buy G729? How many channels we get for 2.95 Euro? How many channels should we buy to support our needs. The cost will be = (2.95 * nr of channels)?

I've just contracted the SOHO account and only need 1 x G.729 channel for 1 x Extension. Remaining extensions (04 in total) will work with G.711. According to this, Do I really need to buy 4 x G.729 channels in order to not leave my extensions without service?